This invention relates to decoding of compressed audio stream data and particularly relates to allocating memory for the decoding of such data.
Decoding compressed audio stream data frequently involves removing jitter from the compressed data. Removing the jitter requires implementation of a jitter buffer, which stores an amount of compressed data, such as compressed packet voice data. Packets typically are put into the jitter buffer from a packet network at a non-constant rate (i.e., with jitter). Data is extracted from the buffer at a constant rate and played out into the telephone network. Previous decoders have sized the jitter buffer to hold a given amount of G.711 data (pulse code modulated (PCM) data transmitted at 64 kilo bits per second(kbps)). This jitter buffer size is usually expressed in bytes or the time duration of G.711 samples. The jitter buffer consumes a significant amount of memory. For example, a 200 millisecond (ms) jitter buffer requires at least 1600 bytes of memory (i.e. 200 ms of G.711 data requires 1600 bytes). Previous decoders do not have the means of re-sizing a jitter buffer whenever the voice decoder algorithm changes dynamically, based on the type of voice decoder algorithm in use. As a result, prior decoders have wasted memory and required excessively large memories. This invention addresses the problem and provides a solution.
Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with the present invention as set forth in the remainder of the present application with reference to the drawings.